[asterisk-users] problem with asterisk - calls where both sidescannot hear each other

[Jorge Mendoza]

Singer,
Assuming that you have no issues with firewalls in the path regarding 
the rtp ports, or hardware/firmware problems, take a look at this patch:
http://www.sineapps.com/news.php?rssid=1019
Please take note if * does not receive rtp packets for any reason, it 
does not send either.

Jorge

Singer Wang wrote:
> I have bindaddr=0.0.0.0 in my sip.conf; what my major problem is that it
> only happens 5-8% of the time.. 
>
> On Wed, 2006-12-06 at 09:56 -0500, Ed Nuñez wrote:
>   
>> If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf
>>
>> Example
>>
>> bindaddr=0.0.0.0
>>
>> will allow SIP traffic on any of your interfaces.
>>
>>
>>
>> Ed Nuñez
>> IT/Telecom Engineer
>>  
>> 4037 Metric Drive
>> Winter Park, FL
>>  
>> (o) 407-384-4200 x 1656
>> (f) 407-384-4222
>> (c) 732-925-0730
>> -----Original Message-----
>> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Singer Wang
>> Sent: Tuesday, December 05, 2006 4:23 PM
>> To: asterisk-users at lists.digium.com
>> Subject: [asterisk-users] problem with asterisk - calls where both sidescannot hear each other
>>
>> Hi,
>>
>> I'm looking for some help with a problem in Asterisk (possibly), and I'm
>> confused as heck what is going on. I've updated to the latest Asterisk
>> version and the problem is still occur. My setup is as follows:
>>
>> I've got Asterisk running on a high end Pentium-IV box running Linux
>> serving 5 calls, it is located in Canada. The calls come in via analog
>> lines through TDM400P cards to Asterisk box, which then converts it to
>> G729 channels to a call center in India over the Internet. Connection
>> between the Asterisk Server and the India call center is done via two
>> Cisco PIX501 devices, The call center in India is running 5 agents using
>> PolyCom phones, and we're using G729 to save bandwith. And yes, we
>> purchused 5 licenses of G729 codec.
>>
>> We're using SIP and a ring all strategy, with the first agent that picks
>> up getting the call. The problem we're having is that about 5-10% calls
>> are not connecting properly. In that both sides can talk but do not hear
>> each other. Since we have recording in step s,5 (in the configuration
>> below), I can verify that it is happening. In these problematic calls,
>> both sides of the call talk but they cannot hear the other side at all.
>>
>> I've gone through most of the documentation and spend hours on Google
>> search, does anyone have any idea what could be the problem? I'm willing
>> to provide more information if asked. 
>>
>>
>> My extensions configuration is roughly the following:
>>
>> [opened]
>> exten => s,1,SetVar(LOOP=1)
>> exten => s,2,Answer
>> exten => s,3,Wait(1)
>> exten => s,4,Background(open-hiq)
>> exten =>
>> s,5,SetVar(MONITOR_FILENAME=/var/spool/asterisk/monitor/inbound/SUPPORT-${UNIQUEID})
>> exten => s,6,Queue(support||||3600)
>> exten => s,7,Voicemail(100|us)
>>
>> exten => 1,1,Goto(opened,s,6)
>>
>> exten => 500,1,Voicemail(500)
>>
>>
>> thanks,
>> Singer Wang
>>
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